On 11-03-08 08:24, Gadi Oron wrote:
> I am trying to record sound on an old Thinkpad T22 that uses the cs46xx
> sound driver.
>
> Each time you start to record you have a 10% chance of having the
> recording completely distorted and having a metallic sound. When you
> look at the waveform it looks as though there are small segments with
> sharp transitions between them, a little like if these segments were
> moved a little from their correct place.
No insights, but I confirm the bug with a TerraTec DMX XFire 1024 (CS4624).
Rene.

Hi, I started to play with ALSA 2 days ago and Im trying to figure out how
ALSA works and how to add PCM devices in the configuration file.
I wrote a simple plugin for my microphone which is supposed to convert the
data into a 32 bps format. Here is how it looks like :
pcm.jcb-in-1 {
type hw
card 0
device 2
}
ctl.jcb-in-1 {
type hw
card 0
}
pcm.MicPlug
{
type plug
slave
{
pcm jcb-in-1
format S32_LE
}
}
So now if Im opening the PCM device named MicPlug and start reading on it, Im
expecting (which may totally be wrong) that the data I read will be in a
32bps format, which is never happening. In fact, the data I read is exactly
in the format that I set using "snd_pcm_hw_params_set_format". Is the
conversion supposed to happen or am I expecting something that is totally
wrong?
Also is there any good documentation about how the asound.conf file works? I
found some stuff on the web but it rather sucks.
Thank you!
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On Tuesday 11 March 2008 06:33, John Sigler wrote:
> I must say that the ALSA
> configuration files look like pure voodoo magic to me.
Me too. I really wish someone would write "Ultimate ALSA Configuration for the
Complete Idiot." It would be a book I would purchase.
--
Darrell Bellerive

> Sadly, I've already tried all of these to no avail.
What recording application are you using? I've had issues where audacity would give me that metalic sound and ardour+jackd would not. And vice versa. Depending on versions and whatnot.
Beyond that I really can't offer any more insight without additional info. Like alsa version, kernel version, contents of /proc/asound/cards, .asoundrc, and whatever else might apply. Aside from upgrading to the latest kernel and latest version of alsa. It might be a known and already fixed issue.
For audacity, most times I end up compiling it from source with the --with-portaudio=v19 parameter to work around various issues. Although it looks like debian caught on and now supplies a version with that option.
HTH

Hi James, thank you for the reply.
>> Each time you start to record you have a
>> 10% chance of having the recording completely
>> distorted and having a metallic sound.
>
> I know that sound. And it is quite ugly. On my snd-hda-intel
board(nVidia MCP61), I have to increase the number of periods to overcome
this sound. default of 2 > > increased to 3 and all was fine, in jackd -n
3. For arecord you might look at different than default buffer/period
sizes. It's basically a latency issue.
>
> Other considerations are to give the audio group realtime permissions so
things like ethernet traffic doesn't cause clicks and other distortions in
the sound.
>/etc/security/limits.conf
Sadly, I've already tried all of these to no avail.
The only thing I can't do is to have the soundcard have it's own IRQ - I
allways get "yenta" together with it.
Someone knows how to disable it or change it's IRQ?
Ciao

On Tue, 11 Mar 2008 07:29:11 -0700
Roger Pryor <rpryor@...> wrote:
> Hi:
>
> I seem to have a problem with Alsa 1.0.16 when compiling.
>
> My system is: an Open SUSE 10.2 running on a Intel DP965LT mobo, with an
> Intel Core2 Duo E6420 processor. 2 G Ram. Intel HDA sound card, which
> has given me lot of problems.
>
> OpenSUSE 10.2 comes with Alsa 10.0.14a, which does not seem to properly
> support the Intel HDA chipset. So, I wanted to upgrade to Alsa 1.0.16. I
> downloaded all available packages from the Alsa site, built and installed
> the driver (using --with-suse=yes), ran alsaconf and rebooted. So far so
> good. I built the library, and installed that. I try to build the utils,
> but it fails with the error "no TLV support in alsa-lib". I check the date
> on /usr/lib64/libasound.so.2.0.0, only to find it was NOT updated with the
> libray installation, yet the version in /usr/lib was updated. Temporarily
> symlinking libasound.2.0.0 in /usr/lib64 to /usr/lib allows the
> compilation of alsa-utils to proceed and complete without error
The symlinking is senseless in this case - /usr/lib64 is meant to be populated
by 64 bit .so files while /usr/lib by 32 bit ones.
Applications complain because they encounter wrong (64 <-> 32) files.
The rest will hopefully be addressed by ALSA developers.
Regards,
Sergei.

Hi:
I seem to have a problem with Alsa 1.0.16 when compiling.
My system is: an Open SUSE 10.2 running on a Intel DP965LT mobo, with an
Intel Core2 Duo E6420 processor. 2 G Ram. Intel HDA sound card, which
has given me lot of problems.
OpenSUSE 10.2 comes with Alsa 10.0.14a, which does not seem to properly
support the Intel HDA chipset. So, I wanted to upgrade to Alsa 1.0.16. I
downloaded all available packages from the Alsa site, built and installed
the driver (using --with-suse=yes), ran alsaconf and rebooted. So far so
good. I built the library, and installed that. I try to build the utils,
but it fails with the error "no TLV support in alsa-lib". I check the date
on /usr/lib64/libasound.so.2.0.0, only to find it was NOT updated with the
libray installation, yet the version in /usr/lib was updated. Temporarily
symlinking libasound.2.0.0 in /usr/lib64 to /usr/lib allows the
compilation of alsa-utils to proceed and complete without error. BUT, all
other applications that use /usr/lib64/libasound now complain about "wrong
ELF type, ELFLIB64", HuH???
Not being a real programmer (Hardware engineer, retired), this says to me
that:
a) Perhaps the installation of alsa-lib is placing the library in
the wrong directory on 64 bit systems.
b) If the ELF type in the /usr/lib64 directory IS 64 bit, why is
that causing a complaint?
At that point, my head aches and I need some help and guidance, please.
--------------------------------------------------------------------------
Roger Pryor Email: rjpryor@...
Vancouver, Canada

Hello Jaroslav,
Jaroslav Kysela wrote:
> John Sigler wrote:
>
>> I have an RME AES-32 PCI board which provides 4 stereo input channels
>> and 4 stereo output channels.
>>
>> (I'm using the hsdpm driver at this time.)
>>
>> I want to use one process per channel, i.e. process A handles stereo
>> input #1 (on the XLR connector #1), process B handles stereo input #2,
>> etc.
>>
>> The processes are independent, in that process A might be started, and
>> only several hours later, process B is started, then a few hours later
>> process A is killed and restarted.
>>
>> Is it possible to do that with the ALSA library?
>
> Yes, look for the dsnoop plugin configuration in alsa-lib.
Thanks for the tip. I will investigate. I must say that the ALSA
configuration files look like pure voodoo magic to me.
I found the following documentation:
doc/asoundrc.txt
http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.htmlhttp://alsa.opensrc.org/.asoundrchttp://alsa.opensrc.org/Dsnoophttp://alsa.opensrc.org/AlsaTips
Is there any other good documentation?
How can alsa-lib open a device node several times when the driver
only allows one process to open it?
I see references to "ipc" in alsa.conf.
My embedded kernel is compiled with
# CONFIG_SYSVIPC is not set
Does the dsnoop plugin require kernel support?
--
Regards.

> >> To use your second card, you have to specifify the device somewhere. If
> >> you want just the hardware abilities, just use the "hw:1" device instead
> >> of "default:1" or whatever you are using now.
> >
> > Using hw:1:0, I get the error message "sample format non available" most
> > of the time, I guess the reason is that it is a 24bit sound card and my
> > files are 16bit.
> >
> > So what I want is:
> > No dmix, but automatic conversion to 24 bit.
> >
> > Is there a simple way to do that?
>
> "plughw:1" should do exactly what you want from it, it seems.
Yes, I was not aware of plughw.
Thanks for the hint.
> >> (your "default" PCM is defined in $PREFIX/share/alsa/cards/ICE1712.conf)
> >
> > On my system it is /usr/share/alsa/cards/ICE1712.conf.
> > Should I really edit that file? I have some hesitations to touch files in
> > the /usr/-directory
>
> No, that was just showing you were things are. If you'd really want to
> change things you'd redefine the default device in /etc/asound.conf or
> ~/.asoundrc
ok, thank you.
~Michael

> Each time you start to record you have a
> 10% chance of having the recording completely
> distorted and having a metallic sound.
I know that sound. And it is quite ugly. On my snd-hda-intel board(nVidia MCP61), I have to increase the number of periods to overcome this sound. default of 2 increased to 3 and all was fine, in jackd -n 3. For arecord you might look at different than default buffer/period sizes. It's basically a latency issue.
Other considerations are to give the audio group realtime permissions so things like ethernet traffic doesn't cause clicks and other distortions in the sound. /etc/security/limits.conf
HTH,
James

hascii wrote:
> the folder /proc/asound/card0/pcm0p is not there any more.
What is the output of the following command?
gunzip - < /proc/config.gz | grep CONFIG_SND_VERBOSE
You might need to enable CONFIG_SND_VERBOSE_PROCFS in your
kernel configuration.
cf. http://alsa.opensrc.org/Proc_asound_documentation
Regards.

Hi Romeo!
I don't have neither. But I have an M-Audio Delta 1010LT, which I suppose is
RELATIVELY close to the revolution 5.1. If they also use the ICE-chip, then
they are nice. I also remember, that we had a lot of traffic about the
revolution cards here. Since I didn't notice those mails for a while, I think
it's a good one.
That all sounds very vague, but still... Did you take a look at the alsa
supported cards? If so and if the Revolution pops up there, then I'd say: Have
a go at it!
My Delta has very nice quality, very reasonable lowlatency and the
BIT-depth/samplingrate should meet your requirements perfectly. Btw.: The
reasonable lowlatency is mostly due to my system. A better PC than mine
(1.8GHz CPU, average Harddisk, DDR-RAM) should really get you 64-128 periods
latency with JACK.
HTH.
Kindest regards
Julien
--------
Music was my first love and it will be my last (John Miles)
======== FIND MY WEB-PROJECT AT: ========
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: =======
http://www.juliencoder.de

Hi everyone,
I ran over a lot of Internet search and could not find any clue to this
issue.
I am trying to record sound on an old Thinkpad T22 that uses the cs46xx
sound driver.
Each time you start to record you have a 10% chance of having the recording
completely distorted and having a metallic sound. When you look at the
waveform it looks as though there are small segments with sharp transitions
between them, a little like if these segments were moved a little from their
correct place.
I've tried many many combination and none gives a solution, including
- Native OSS, 3.92, 4.0
- kernel 2.4, 2.6.11, 2.6.26 (rh7.3, fedora 1,2,7,8)
- Knoppix live CD
- Activate thinkpad=1 for the module
- Using native ALSA or OSS emulation.
- Changing fragment size
- Resetting mixer setting before recording
- Changing recording format LE,BE, 8bit, 16bit
- Changing recording frequency
- Disable ACPI
- Change IRQ of the sound-card
- Using a low-latency kernel and giving the recording process
real-time priority (FIFO_SCHED)
None gave an improvement. It works fine on the Win2k system that is
installed on the same computer.
Does anyone have any clue to what might be going on?
Is there any way to do a hard reset to the sound card (without
unloading-reloading the modules)?
Thank you for the help.

I am looking for a well supported PCI sound card under ALSA with some
characteristics: 24 bits, 48 Khz (192 Khz would be a plus, 24-bit/96kHz
mic/line recording would also be nice) and a S/PDF coaxial or optical
output. I use some music software(like real time guitar effects) and jack,
so I need some low latency. I also have 2.0 speakers (I don't care about
surround sound) with S/PDF out (optical and coaxial).
I have a low budget -around $100- and I have two sound cards in mind, which
are Audiotrak prodigy HD2 and M-Audio Revolution 5.1
Can somebody with this hardware give me feedback on these cards??
Or maybe another brand or model?
by the way I use debian sid.
Thanks for the help
<http://www.taligentx.com/projects/opticalconverter/&gt;